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qemu/audio/dsoundaudio.c
ths 4fddf62a74 Use WIN32_LEAN_AND_MEAN, by Stefan Weil. 
git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@3832 c046a42c-6fe2-441c-8c8c-71466251a162
2007-12-17 04:42:29 +00:00

1084 lines
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/*
* QEMU DirectSound audio driver
*
* Copyright (c) 2005 Vassili Karpov (malc)
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/*
* SEAL 1.07 by Carlos 'pel' Hasan was used as documentation
*/
#include "qemu-common.h"
#include "audio.h"
#define AUDIO_CAP "dsound"
#include "audio_int.h"
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
#include <mmsystem.h>
#include <objbase.h>
#include <dsound.h>
/* #define DEBUG_DSOUND */
static struct {
int lock_retries;
int restore_retries;
int getstatus_retries;
int set_primary;
int bufsize_in;
int bufsize_out;
audsettings_t settings;
int latency_millis;
} conf = {
1,
1,
1,
0,
16384,
16384,
{
44100,
2,
AUD_FMT_S16
},
10
};
typedef struct {
LPDIRECTSOUND dsound;
LPDIRECTSOUNDCAPTURE dsound_capture;
LPDIRECTSOUNDBUFFER dsound_primary_buffer;
audsettings_t settings;
} dsound;
static dsound glob_dsound;
typedef struct {
HWVoiceOut hw;
LPDIRECTSOUNDBUFFER dsound_buffer;
DWORD old_pos;
int first_time;
#ifdef DEBUG_DSOUND
DWORD old_ppos;
DWORD played;
DWORD mixed;
#endif
} DSoundVoiceOut;
typedef struct {
HWVoiceIn hw;
int first_time;
LPDIRECTSOUNDCAPTUREBUFFER dsound_capture_buffer;
} DSoundVoiceIn;
static void dsound_log_hresult (HRESULT hr)
{
const char *str = "BUG";
switch (hr) {
case DS_OK:
str = "The method succeeded";
break;
#ifdef DS_NO_VIRTUALIZATION
case DS_NO_VIRTUALIZATION:
str = "The buffer was created, but another 3D algorithm was substituted";
break;
#endif
#ifdef DS_INCOMPLETE
case DS_INCOMPLETE:
str = "The method succeeded, but not all the optional effects were obtained";
break;
#endif
#ifdef DSERR_ACCESSDENIED
case DSERR_ACCESSDENIED:
str = "The request failed because access was denied";
break;
#endif
#ifdef DSERR_ALLOCATED
case DSERR_ALLOCATED:
str = "The request failed because resources, such as a priority level, were already in use by another caller";
break;
#endif
#ifdef DSERR_ALREADYINITIALIZED
case DSERR_ALREADYINITIALIZED:
str = "The object is already initialized";
break;
#endif
#ifdef DSERR_BADFORMAT
case DSERR_BADFORMAT:
str = "The specified wave format is not supported";
break;
#endif
#ifdef DSERR_BADSENDBUFFERGUID
case DSERR_BADSENDBUFFERGUID:
str = "The GUID specified in an audiopath file does not match a valid mix-in buffer";
break;
#endif
#ifdef DSERR_BUFFERLOST
case DSERR_BUFFERLOST:
str = "The buffer memory has been lost and must be restored";
break;
#endif
#ifdef DSERR_BUFFERTOOSMALL
case DSERR_BUFFERTOOSMALL:
str = "The buffer size is not great enough to enable effects processing";
break;
#endif
#ifdef DSERR_CONTROLUNAVAIL
case DSERR_CONTROLUNAVAIL:
str = "The buffer control (volume, pan, and so on) requested by the caller is not available. Controls must be specified when the buffer is created, using the dwFlags member of DSBUFFERDESC";
break;
#endif
#ifdef DSERR_DS8_REQUIRED
case DSERR_DS8_REQUIRED:
str = "A DirectSound object of class CLSID_DirectSound8 or later is required for the requested functionality. For more information, see IDirectSound8 Interface";
break;
#endif
#ifdef DSERR_FXUNAVAILABLE
case DSERR_FXUNAVAILABLE:
str = "The effects requested could not be found on the system, or they are in the wrong order or in the wrong location; for example, an effect expected in hardware was found in software";
break;
#endif
#ifdef DSERR_GENERIC
case DSERR_GENERIC :
str = "An undetermined error occurred inside the DirectSound subsystem";
break;
#endif
#ifdef DSERR_INVALIDCALL
case DSERR_INVALIDCALL:
str = "This function is not valid for the current state of this object";
break;
#endif
#ifdef DSERR_INVALIDPARAM
case DSERR_INVALIDPARAM:
str = "An invalid parameter was passed to the returning function";
break;
#endif
#ifdef DSERR_NOAGGREGATION
case DSERR_NOAGGREGATION:
str = "The object does not support aggregation";
break;
#endif
#ifdef DSERR_NODRIVER
case DSERR_NODRIVER:
str = "No sound driver is available for use, or the given GUID is not a valid DirectSound device ID";
break;
#endif
#ifdef DSERR_NOINTERFACE
case DSERR_NOINTERFACE:
str = "The requested COM interface is not available";
break;
#endif
#ifdef DSERR_OBJECTNOTFOUND
case DSERR_OBJECTNOTFOUND:
str = "The requested object was not found";
break;
#endif
#ifdef DSERR_OTHERAPPHASPRIO
case DSERR_OTHERAPPHASPRIO:
str = "Another application has a higher priority level, preventing this call from succeeding";
break;
#endif
#ifdef DSERR_OUTOFMEMORY
case DSERR_OUTOFMEMORY:
str = "The DirectSound subsystem could not allocate sufficient memory to complete the caller's request";
break;
#endif
#ifdef DSERR_PRIOLEVELNEEDED
case DSERR_PRIOLEVELNEEDED:
str = "A cooperative level of DSSCL_PRIORITY or higher is required";
break;
#endif
#ifdef DSERR_SENDLOOP
case DSERR_SENDLOOP:
str = "A circular loop of send effects was detected";
break;
#endif
#ifdef DSERR_UNINITIALIZED
case DSERR_UNINITIALIZED:
str = "The Initialize method has not been called or has not been called successfully before other methods were called";
break;
#endif
#ifdef DSERR_UNSUPPORTED
case DSERR_UNSUPPORTED:
str = "The function called is not supported at this time";
break;
#endif
default:
AUD_log (AUDIO_CAP, "Reason: Unknown (HRESULT %#lx)\n", hr);
return;
}
AUD_log (AUDIO_CAP, "Reason: %s\n", str);
}
static void GCC_FMT_ATTR (2, 3) dsound_logerr (
HRESULT hr,
const char *fmt,
...
)
{
va_list ap;
va_start (ap, fmt);
AUD_vlog (AUDIO_CAP, fmt, ap);
va_end (ap);
dsound_log_hresult (hr);
}
static void GCC_FMT_ATTR (3, 4) dsound_logerr2 (
HRESULT hr,
const char *typ,
const char *fmt,
...
)
{
va_list ap;
AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
va_start (ap, fmt);
AUD_vlog (AUDIO_CAP, fmt, ap);
va_end (ap);
dsound_log_hresult (hr);
}
static DWORD millis_to_bytes (struct audio_pcm_info *info, DWORD millis)
{
return (millis * info->bytes_per_second) / 1000;
}
#ifdef DEBUG_DSOUND
static void print_wave_format (WAVEFORMATEX *wfx)
{
dolog ("tag = %d\n", wfx->wFormatTag);
dolog ("nChannels = %d\n", wfx->nChannels);
dolog ("nSamplesPerSec = %ld\n", wfx->nSamplesPerSec);
dolog ("nAvgBytesPerSec = %ld\n", wfx->nAvgBytesPerSec);
dolog ("nBlockAlign = %d\n", wfx->nBlockAlign);
dolog ("wBitsPerSample = %d\n", wfx->wBitsPerSample);
dolog ("cbSize = %d\n", wfx->cbSize);
}
#endif
static int dsound_restore_out (LPDIRECTSOUNDBUFFER dsb)
{
HRESULT hr;
int i;
for (i = 0; i < conf.restore_retries; ++i) {
hr = IDirectSoundBuffer_Restore (dsb);
switch (hr) {
case DS_OK:
return 0;
case DSERR_BUFFERLOST:
continue;
default:
dsound_logerr (hr, "Could not restore playback buffer\n");
return -1;
}
}
dolog ("%d attempts to restore playback buffer failed\n", i);
return -1;
}
static int waveformat_from_audio_settings (WAVEFORMATEX *wfx, audsettings_t *as)
{
memset (wfx, 0, sizeof (*wfx));
wfx->wFormatTag = WAVE_FORMAT_PCM;
wfx->nChannels = as->nchannels;
wfx->nSamplesPerSec = as->freq;
wfx->nAvgBytesPerSec = as->freq << (as->nchannels == 2);
wfx->nBlockAlign = 1 << (as->nchannels == 2);
wfx->cbSize = 0;
switch (as->fmt) {
case AUD_FMT_S8:
wfx->wBitsPerSample = 8;
break;
case AUD_FMT_U8:
wfx->wBitsPerSample = 8;
break;
case AUD_FMT_S16:
wfx->wBitsPerSample = 16;
wfx->nAvgBytesPerSec <<= 1;
wfx->nBlockAlign <<= 1;
break;
case AUD_FMT_U16:
wfx->wBitsPerSample = 16;
wfx->nAvgBytesPerSec <<= 1;
wfx->nBlockAlign <<= 1;
break;
default:
dolog ("Internal logic error: Bad audio format %d\n", as->freq);
return -1;
}
return 0;
}
static int waveformat_to_audio_settings (WAVEFORMATEX *wfx, audsettings_t *as)
{
if (wfx->wFormatTag != WAVE_FORMAT_PCM) {
dolog ("Invalid wave format, tag is not PCM, but %d\n",
wfx->wFormatTag);
return -1;
}
if (!wfx->nSamplesPerSec) {
dolog ("Invalid wave format, frequency is zero\n");
return -1;
}
as->freq = wfx->nSamplesPerSec;
switch (wfx->nChannels) {
case 1:
as->nchannels = 1;
break;
case 2:
as->nchannels = 2;
break;
default:
dolog (
"Invalid wave format, number of channels is not 1 or 2, but %d\n",
wfx->nChannels
);
return -1;
}
switch (wfx->wBitsPerSample) {
case 8:
as->fmt = AUD_FMT_U8;
break;
case 16:
as->fmt = AUD_FMT_S16;
break;
default:
dolog ("Invalid wave format, bits per sample is not 8 or 16, but %d\n",
wfx->wBitsPerSample);
return -1;
}
return 0;
}
#include "dsound_template.h"
#define DSBTYPE_IN
#include "dsound_template.h"
#undef DSBTYPE_IN
static int dsound_get_status_out (LPDIRECTSOUNDBUFFER dsb, DWORD *statusp)
{
HRESULT hr;
int i;
for (i = 0; i < conf.getstatus_retries; ++i) {
hr = IDirectSoundBuffer_GetStatus (dsb, statusp);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not get playback buffer status\n");
return -1;
}
if (*statusp & DSERR_BUFFERLOST) {
if (dsound_restore_out (dsb)) {
return -1;
}
continue;
}
break;
}
return 0;
}
static int dsound_get_status_in (LPDIRECTSOUNDCAPTUREBUFFER dscb,
DWORD *statusp)
{
HRESULT hr;
hr = IDirectSoundCaptureBuffer_GetStatus (dscb, statusp);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not get capture buffer status\n");
return -1;
}
return 0;
}
static void dsound_write_sample (HWVoiceOut *hw, uint8_t *dst, int dst_len)
{
int src_len1 = dst_len;
int src_len2 = 0;
int pos = hw->rpos + dst_len;
st_sample_t *src1 = hw->mix_buf + hw->rpos;
st_sample_t *src2 = NULL;
if (pos > hw->samples) {
src_len1 = hw->samples - hw->rpos;
src2 = hw->mix_buf;
src_len2 = dst_len - src_len1;
pos = src_len2;
}
if (src_len1) {
hw->clip (dst, src1, src_len1);
}
if (src_len2) {
dst = advance (dst, src_len1 << hw->info.shift);
hw->clip (dst, src2, src_len2);
}
hw->rpos = pos % hw->samples;
}
static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb)
{
int err;
LPVOID p1, p2;
DWORD blen1, blen2, len1, len2;
err = dsound_lock_out (
dsb,
&hw->info,
0,
hw->samples << hw->info.shift,
&p1, &p2,
&blen1, &blen2,
1
);
if (err) {
return;
}
len1 = blen1 >> hw->info.shift;
len2 = blen2 >> hw->info.shift;
#ifdef DEBUG_DSOUND
dolog ("clear %p,%ld,%ld %p,%ld,%ld\n",
p1, blen1, len1,
p2, blen2, len2);
#endif
if (p1 && len1) {
audio_pcm_info_clear_buf (&hw->info, p1, len1);
}
if (p2 && len2) {
audio_pcm_info_clear_buf (&hw->info, p2, len2);
}
dsound_unlock_out (dsb, p1, p2, blen1, blen2);
}
static void dsound_close (dsound *s)
{
HRESULT hr;
if (s->dsound_primary_buffer) {
hr = IDirectSoundBuffer_Release (s->dsound_primary_buffer);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not release primary buffer\n");
}
s->dsound_primary_buffer = NULL;
}
}
static int dsound_open (dsound *s)
{
int err;
HRESULT hr;
WAVEFORMATEX wfx;
DSBUFFERDESC dsbd;
HWND hwnd;
hwnd = GetForegroundWindow ();
hr = IDirectSound_SetCooperativeLevel (
s->dsound,
hwnd,
DSSCL_PRIORITY
);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not set cooperative level for window %p\n",
hwnd);
return -1;
}
if (!conf.set_primary) {
return 0;
}
err = waveformat_from_audio_settings (&wfx, &conf.settings);
if (err) {
return -1;
}
memset (&dsbd, 0, sizeof (dsbd));
dsbd.dwSize = sizeof (dsbd);
dsbd.dwFlags = DSBCAPS_PRIMARYBUFFER;
dsbd.dwBufferBytes = 0;
dsbd.lpwfxFormat = NULL;
hr = IDirectSound_CreateSoundBuffer (
s->dsound,
&dsbd,
&s->dsound_primary_buffer,
NULL
);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not create primary playback buffer\n");
return -1;
}
hr = IDirectSoundBuffer_SetFormat (s->dsound_primary_buffer, &wfx);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not set primary playback buffer format\n");
}
hr = IDirectSoundBuffer_GetFormat (
s->dsound_primary_buffer,
&wfx,
sizeof (wfx),
NULL
);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not get primary playback buffer format\n");
goto fail0;
}
#ifdef DEBUG_DSOUND
dolog ("Primary\n");
print_wave_format (&wfx);
#endif
err = waveformat_to_audio_settings (&wfx, &s->settings);
if (err) {
goto fail0;
}
return 0;
fail0:
dsound_close (s);
return -1;
}
static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
HRESULT hr;
DWORD status;
DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;
if (!dsb) {
dolog ("Attempt to control voice without a buffer\n");
return 0;
}
switch (cmd) {
case VOICE_ENABLE:
if (dsound_get_status_out (dsb, &status)) {
return -1;
}
if (status & DSBSTATUS_PLAYING) {
dolog ("warning: Voice is already playing\n");
return 0;
}
dsound_clear_sample (hw, dsb);
hr = IDirectSoundBuffer_Play (dsb, 0, 0, DSBPLAY_LOOPING);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not start playing buffer\n");
return -1;
}
break;
case VOICE_DISABLE:
if (dsound_get_status_out (dsb, &status)) {
return -1;
}
if (status & DSBSTATUS_PLAYING) {
hr = IDirectSoundBuffer_Stop (dsb);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not stop playing buffer\n");
return -1;
}
}
else {
dolog ("warning: Voice is not playing\n");
}
break;
}
return 0;
}
static int dsound_write (SWVoiceOut *sw, void *buf, int len)
{
return audio_pcm_sw_write (sw, buf, len);
}
static int dsound_run_out (HWVoiceOut *hw)
{
int err;
HRESULT hr;
DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;
int live, len, hwshift;
DWORD blen1, blen2;
DWORD len1, len2;
DWORD decr;
DWORD wpos, ppos, old_pos;
LPVOID p1, p2;
int bufsize;
if (!dsb) {
dolog ("Attempt to run empty with playback buffer\n");
return 0;
}
hwshift = hw->info.shift;
bufsize = hw->samples << hwshift;
live = audio_pcm_hw_get_live_out (hw);
hr = IDirectSoundBuffer_GetCurrentPosition (
dsb,
&ppos,
ds->first_time ? &wpos : NULL
);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not get playback buffer position\n");
return 0;
}
len = live << hwshift;
if (ds->first_time) {
if (conf.latency_millis) {
DWORD cur_blat;
cur_blat = audio_ring_dist (wpos, ppos, bufsize);
ds->first_time = 0;
old_pos = wpos;
old_pos +=
millis_to_bytes (&hw->info, conf.latency_millis) - cur_blat;
old_pos %= bufsize;
old_pos &= ~hw->info.align;
}
else {
old_pos = wpos;
}
#ifdef DEBUG_DSOUND
ds->played = 0;
ds->mixed = 0;
#endif
}
else {
if (ds->old_pos == ppos) {
#ifdef DEBUG_DSOUND
dolog ("old_pos == ppos\n");
#endif
return 0;
}
#ifdef DEBUG_DSOUND
ds->played += audio_ring_dist (ds->old_pos, ppos, hw->bufsize);
#endif
old_pos = ds->old_pos;
}
if ((old_pos < ppos) && ((old_pos + len) > ppos)) {
len = ppos - old_pos;
}
else {
if ((old_pos > ppos) && ((old_pos + len) > (ppos + bufsize))) {
len = bufsize - old_pos + ppos;
}
}
if (audio_bug (AUDIO_FUNC, len < 0 || len > bufsize)) {
dolog ("len=%d bufsize=%d old_pos=%ld ppos=%ld\n",
len, bufsize, old_pos, ppos);
return 0;
}
len &= ~hw->info.align;
if (!len) {
return 0;
}
#ifdef DEBUG_DSOUND
ds->old_ppos = ppos;
#endif
err = dsound_lock_out (
dsb,
&hw->info,
old_pos,
len,
&p1, &p2,
&blen1, &blen2,
0
);
if (err) {
return 0;
}
len1 = blen1 >> hwshift;
len2 = blen2 >> hwshift;
decr = len1 + len2;
if (p1 && len1) {
dsound_write_sample (hw, p1, len1);
}
if (p2 && len2) {
dsound_write_sample (hw, p2, len2);
}
dsound_unlock_out (dsb, p1, p2, blen1, blen2);
ds->old_pos = (old_pos + (decr << hwshift)) % bufsize;
#ifdef DEBUG_DSOUND
ds->mixed += decr << hwshift;
dolog ("played %lu mixed %lu diff %ld sec %f\n",
ds->played,
ds->mixed,
ds->mixed - ds->played,
abs (ds->mixed - ds->played) / (double) hw->info.bytes_per_second);
#endif
return decr;
}
static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
HRESULT hr;
DWORD status;
DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
LPDIRECTSOUNDCAPTUREBUFFER dscb = ds->dsound_capture_buffer;
if (!dscb) {
dolog ("Attempt to control capture voice without a buffer\n");
return -1;
}
switch (cmd) {
case VOICE_ENABLE:
if (dsound_get_status_in (dscb, &status)) {
return -1;
}
if (status & DSCBSTATUS_CAPTURING) {
dolog ("warning: Voice is already capturing\n");
return 0;
}
/* clear ?? */
hr = IDirectSoundCaptureBuffer_Start (dscb, DSCBSTART_LOOPING);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not start capturing\n");
return -1;
}
break;
case VOICE_DISABLE:
if (dsound_get_status_in (dscb, &status)) {
return -1;
}
if (status & DSCBSTATUS_CAPTURING) {
hr = IDirectSoundCaptureBuffer_Stop (dscb);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not stop capturing\n");
return -1;
}
}
else {
dolog ("warning: Voice is not capturing\n");
}
break;
}
return 0;
}
static int dsound_read (SWVoiceIn *sw, void *buf, int len)
{
return audio_pcm_sw_read (sw, buf, len);
}
static int dsound_run_in (HWVoiceIn *hw)
{
int err;
HRESULT hr;
DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
LPDIRECTSOUNDCAPTUREBUFFER dscb = ds->dsound_capture_buffer;
int live, len, dead;
DWORD blen1, blen2;
DWORD len1, len2;
DWORD decr;
DWORD cpos, rpos;
LPVOID p1, p2;
int hwshift;
if (!dscb) {
dolog ("Attempt to run without capture buffer\n");
return 0;
}
hwshift = hw->info.shift;
live = audio_pcm_hw_get_live_in (hw);
dead = hw->samples - live;
if (!dead) {
return 0;
}
hr = IDirectSoundCaptureBuffer_GetCurrentPosition (
dscb,
&cpos,
ds->first_time ? &rpos : NULL
);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not get capture buffer position\n");
return 0;
}
if (ds->first_time) {
ds->first_time = 0;
if (rpos & hw->info.align) {
ldebug ("warning: Misaligned capture read position %ld(%d)\n",
rpos, hw->info.align);
}
hw->wpos = rpos >> hwshift;
}
if (cpos & hw->info.align) {
ldebug ("warning: Misaligned capture position %ld(%d)\n",
cpos, hw->info.align);
}
cpos >>= hwshift;
len = audio_ring_dist (cpos, hw->wpos, hw->samples);
if (!len) {
return 0;
}
len = audio_MIN (len, dead);
err = dsound_lock_in (
dscb,
&hw->info,
hw->wpos << hwshift,
len << hwshift,
&p1,
&p2,
&blen1,
&blen2,
0
);
if (err) {
return 0;
}
len1 = blen1 >> hwshift;
len2 = blen2 >> hwshift;
decr = len1 + len2;
if (p1 && len1) {
hw->conv (hw->conv_buf + hw->wpos, p1, len1, &nominal_volume);
}
if (p2 && len2) {
hw->conv (hw->conv_buf, p2, len2, &nominal_volume);
}
dsound_unlock_in (dscb, p1, p2, blen1, blen2);
hw->wpos = (hw->wpos + decr) % hw->samples;
return decr;
}
static void dsound_audio_fini (void *opaque)
{
HRESULT hr;
dsound *s = opaque;
if (!s->dsound) {
return;
}
hr = IDirectSound_Release (s->dsound);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not release DirectSound\n");
}
s->dsound = NULL;
if (!s->dsound_capture) {
return;
}
hr = IDirectSoundCapture_Release (s->dsound_capture);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not release DirectSoundCapture\n");
}
s->dsound_capture = NULL;
}
static void *dsound_audio_init (void)
{
int err;
HRESULT hr;
dsound *s = &glob_dsound;
hr = CoInitialize (NULL);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not initialize COM\n");
return NULL;
}
hr = CoCreateInstance (
&CLSID_DirectSound,
NULL,
CLSCTX_ALL,
&IID_IDirectSound,
(void **) &s->dsound
);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not create DirectSound instance\n");
return NULL;
}
hr = IDirectSound_Initialize (s->dsound, NULL);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not initialize DirectSound\n");
hr = IDirectSound_Release (s->dsound);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not release DirectSound\n");
}
s->dsound = NULL;
return NULL;
}
hr = CoCreateInstance (
&CLSID_DirectSoundCapture,
NULL,
CLSCTX_ALL,
&IID_IDirectSoundCapture,
(void **) &s->dsound_capture
);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not create DirectSoundCapture instance\n");
}
else {
hr = IDirectSoundCapture_Initialize (s->dsound_capture, NULL);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not initialize DirectSoundCapture\n");
hr = IDirectSoundCapture_Release (s->dsound_capture);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not release DirectSoundCapture\n");
}
s->dsound_capture = NULL;
}
}
err = dsound_open (s);
if (err) {
dsound_audio_fini (s);
return NULL;
}
return s;
}
static struct audio_option dsound_options[] = {
{"LOCK_RETRIES", AUD_OPT_INT, &conf.lock_retries,
"Number of times to attempt locking the buffer", NULL, 0},
{"RESTOURE_RETRIES", AUD_OPT_INT, &conf.restore_retries,
"Number of times to attempt restoring the buffer", NULL, 0},
{"GETSTATUS_RETRIES", AUD_OPT_INT, &conf.getstatus_retries,
"Number of times to attempt getting status of the buffer", NULL, 0},
{"SET_PRIMARY", AUD_OPT_BOOL, &conf.set_primary,
"Set the parameters of primary buffer", NULL, 0},
{"LATENCY_MILLIS", AUD_OPT_INT, &conf.latency_millis,
"(undocumented)", NULL, 0},
{"PRIMARY_FREQ", AUD_OPT_INT, &conf.settings.freq,
"Primary buffer frequency", NULL, 0},
{"PRIMARY_CHANNELS", AUD_OPT_INT, &conf.settings.nchannels,
"Primary buffer number of channels (1 - mono, 2 - stereo)", NULL, 0},
{"PRIMARY_FMT", AUD_OPT_FMT, &conf.settings.fmt,
"Primary buffer format", NULL, 0},
{"BUFSIZE_OUT", AUD_OPT_INT, &conf.bufsize_out,
"(undocumented)", NULL, 0},
{"BUFSIZE_IN", AUD_OPT_INT, &conf.bufsize_in,
"(undocumented)", NULL, 0},
{NULL, 0, NULL, NULL, NULL, 0}
};
static struct audio_pcm_ops dsound_pcm_ops = {
dsound_init_out,
dsound_fini_out,
dsound_run_out,
dsound_write,
dsound_ctl_out,
dsound_init_in,
dsound_fini_in,
dsound_run_in,
dsound_read,
dsound_ctl_in
};
struct audio_driver dsound_audio_driver = {
INIT_FIELD (name = ) "dsound",
INIT_FIELD (descr = )
"DirectSound http://wikipedia.org/wiki/DirectSound",
INIT_FIELD (options = ) dsound_options,
INIT_FIELD (init = ) dsound_audio_init,
INIT_FIELD (fini = ) dsound_audio_fini,
INIT_FIELD (pcm_ops = ) &dsound_pcm_ops,
INIT_FIELD (can_be_default = ) 1,
INIT_FIELD (max_voices_out = ) INT_MAX,
INIT_FIELD (max_voices_in = ) 1,
INIT_FIELD (voice_size_out = ) sizeof (DSoundVoiceOut),
INIT_FIELD (voice_size_in = ) sizeof (DSoundVoiceIn)
};
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